view libao2/ao_alsa5.c @ 23510:a6c619ee9d30

Teletext support for tv:// (v4l and v4l2 only) modified patch from Otvos Attila oattila at chello dot hu Module uses zvbi library for all low-level VBI operations (like I/O with vbi device, converting vbi pages into usefull vbi_page stuctures, rendering them into RGB32 images). All teletext related stuff (except properties, slave commands and rendering osd in text mode or RGB32 rendered teletext pages in spu mode) is implemented in tvi_vbi.c New properties: teletext_page - switching between pages teletext_mode - switch between on/off/opaque/transparent modes teletext_format - (currently read-only) allows to get format info (black/white,gray,text) teletext_half_page - trivial zooming (displaying top/bottom half of teletext page) New slave commands: teletext_add_dec - user interface for jumping to any page by editing page number interactively teletext_go_link - goes though links, specified on current page

/*
  ao_alsa5 - ALSA-0.5.x output plugin for MPlayer

  (C) Alex Beregszaszi

  Thanks to Arpi for helping me ;)
*/

#include <errno.h>
#include <sys/asoundlib.h>

#include "config.h"

#include "audio_out.h"
#include "audio_out_internal.h"
#include "libaf/af_format.h"

#include "mp_msg.h"
#include "help_mp.h"

static ao_info_t info = 
{
    "ALSA-0.5.x audio output",
    "alsa5",
    "Alex Beregszaszi",
    ""
};

LIBAO_EXTERN(alsa5)

static snd_pcm_t *alsa_handler;
static snd_pcm_format_t alsa_format;
static int alsa_rate = SND_PCM_RATE_CONTINUOUS;

/* to set/get/query special features/parameters */
static int control(int cmd, void *arg)
{
    return(CONTROL_UNKNOWN);
}

/*
    open & setup audio device
    return: 1=success 0=fail
*/
static int init(int rate_hz, int channels, int format, int flags)
{
    int err;
    int cards = -1;
    snd_pcm_channel_params_t params;
    snd_pcm_channel_setup_t setup;
    snd_pcm_info_t info;
    snd_pcm_channel_info_t chninfo;

    mp_msg(MSGT_AO, MSGL_INFO, MSGTR_AO_ALSA5_InitInfo, rate_hz,
	channels, af_fmt2str_short(format));

    alsa_handler = NULL;

    mp_msg(MSGT_AO, MSGL_V, "alsa-init: compiled for ALSA-%s (%d)\n", SND_LIB_VERSION_STR,
        SND_LIB_VERSION);

    if ((cards = snd_cards()) < 0)
    {
	mp_msg(MSGT_AO, MSGL_ERR, MSGTR_AO_ALSA5_SoundCardNotFound);
	return(0);
    }

    ao_data.format = format;
    ao_data.channels = channels;
    ao_data.samplerate = rate_hz;
    ao_data.bps = ao_data.samplerate*ao_data.channels;
    ao_data.outburst = OUTBURST;
    ao_data.buffersize = 16384;

    memset(&alsa_format, 0, sizeof(alsa_format));
    switch (format)
    {
	case AF_FORMAT_S8:
	    alsa_format.format = SND_PCM_SFMT_S8;
	    break;
	case AF_FORMAT_U8:
	    alsa_format.format = SND_PCM_SFMT_U8;
	    break;
	case AF_FORMAT_U16_LE:
	    alsa_format.format = SND_PCM_SFMT_U16_LE;
	    break;
	case AF_FORMAT_U16_BE:
	    alsa_format.format = SND_PCM_SFMT_U16_BE;
	    break;
#ifndef WORDS_BIGENDIAN
	case AF_FORMAT_AC3:
#endif
	case AF_FORMAT_S16_LE:
	    alsa_format.format = SND_PCM_SFMT_S16_LE;
	    break;
#ifdef WORDS_BIGENDIAN
	case AF_FORMAT_AC3:
#endif
	case AF_FORMAT_S16_BE:
	    alsa_format.format = SND_PCM_SFMT_S16_BE;
	    break;
	default:
	    alsa_format.format = SND_PCM_SFMT_MPEG;
	    break;
    }
    
    switch(alsa_format.format)
    {
	case SND_PCM_SFMT_S16_LE:
	case SND_PCM_SFMT_U16_LE:
	    ao_data.bps *= 2;
	    break;
	case -1:
	    mp_msg(MSGT_AO, MSGL_ERR, MSGTR_AO_ALSA5_InvalidFormatReq,af_fmt2str_short(format));
	    return(0);
	default:
	    break;
    }

    switch(rate_hz)
    {
	case 8000:
	    alsa_rate = SND_PCM_RATE_8000;
	    break;
	case 11025:
	    alsa_rate = SND_PCM_RATE_11025;
	    break;
	case 16000:
	    alsa_rate = SND_PCM_RATE_16000;
	    break;
	case 22050:
	    alsa_rate = SND_PCM_RATE_22050;
	    break;
	case 32000:
	    alsa_rate = SND_PCM_RATE_32000;
	    break;
	case 44100:
	    alsa_rate = SND_PCM_RATE_44100;
	    break;
	case 48000:
	    alsa_rate = SND_PCM_RATE_48000;
	    break;
	case 88200:
	    alsa_rate = SND_PCM_RATE_88200;
	    break;
	case 96000:
	    alsa_rate = SND_PCM_RATE_96000;
	    break;
	case 176400:
	    alsa_rate = SND_PCM_RATE_176400;
	    break;
	case 192000:
	    alsa_rate = SND_PCM_RATE_192000;
	    break;
	default:
	    alsa_rate = SND_PCM_RATE_CONTINUOUS;
	    break;
    }

    alsa_format.rate = ao_data.samplerate;
    alsa_format.voices = ao_data.channels;
    alsa_format.interleave = 1;

    if ((err = snd_pcm_open(&alsa_handler, 0, 0, SND_PCM_OPEN_PLAYBACK)) < 0)
    {
	mp_msg(MSGT_AO, MSGL_ERR, MSGTR_AO_ALSA5_PlayBackError, snd_strerror(err));
	return(0);
    }

    if ((err = snd_pcm_info(alsa_handler, &info)) < 0)
    {
	mp_msg(MSGT_AO, MSGL_ERR, MSGTR_AO_ALSA5_PcmInfoError, snd_strerror(err));
	return(0);
    }

    mp_msg(MSGT_AO, MSGL_INFO, MSGTR_AO_ALSA5_SoundcardsFound,
	cards, info.name);

    if (info.flags & SND_PCM_INFO_PLAYBACK)
    {
	memset(&chninfo, 0, sizeof(chninfo));
	chninfo.channel = SND_PCM_CHANNEL_PLAYBACK;
	if ((err = snd_pcm_channel_info(alsa_handler, &chninfo)) < 0)
	{
	    mp_msg(MSGT_AO, MSGL_ERR, MSGTR_AO_ALSA5_PcmChanInfoError, snd_strerror(err));
	    return(0);
	}

#ifndef __QNX__
	if (chninfo.buffer_size)
	    ao_data.buffersize = chninfo.buffer_size;
#endif

	mp_msg(MSGT_AO, MSGL_V, "alsa-init: setting preferred buffer size from driver: %d bytes\n",
	    ao_data.buffersize);
    }

    memset(&params, 0, sizeof(params));
    params.channel = SND_PCM_CHANNEL_PLAYBACK;
    params.mode = SND_PCM_MODE_STREAM;
    params.format = alsa_format;
    params.start_mode = SND_PCM_START_DATA;
    params.stop_mode = SND_PCM_STOP_ROLLOVER;
    params.buf.stream.queue_size = ao_data.buffersize;
    params.buf.stream.fill = SND_PCM_FILL_NONE;

    if ((err = snd_pcm_channel_params(alsa_handler, &params)) < 0)
    {
	mp_msg(MSGT_AO, MSGL_ERR, MSGTR_AO_ALSA5_CantSetParms, snd_strerror(err));
	return(0);
    }

    memset(&setup, 0, sizeof(setup));
    setup.channel = SND_PCM_CHANNEL_PLAYBACK;
    setup.mode = SND_PCM_MODE_STREAM;
    setup.format = alsa_format;
    setup.buf.stream.queue_size = ao_data.buffersize;
    setup.msbits_per_sample = ao_data.bps;
    
    if ((err = snd_pcm_channel_setup(alsa_handler, &setup)) < 0)
    {
	mp_msg(MSGT_AO, MSGL_ERR, MSGTR_AO_ALSA5_CantSetChan, snd_strerror(err));
	return(0);
    }

    if ((err = snd_pcm_channel_prepare(alsa_handler, SND_PCM_CHANNEL_PLAYBACK)) < 0)
    {
	mp_msg(MSGT_AO, MSGL_ERR, MSGTR_AO_ALSA5_ChanPrepareError, snd_strerror(err));
	return(0);
    }

    mp_msg(MSGT_AO, MSGL_INFO, "AUDIO: %d Hz/%d channels/%d bps/%d bytes buffer/%s\n",
	ao_data.samplerate, ao_data.channels, ao_data.bps, ao_data.buffersize,
	snd_pcm_get_format_name(alsa_format.format));
    return(1);
}

/* close audio device */
static void uninit(int immed)
{
    int err;

    if ((err = snd_pcm_playback_drain(alsa_handler)) < 0)
    {
	mp_msg(MSGT_AO, MSGL_ERR, MSGTR_AO_ALSA5_DrainError, snd_strerror(err));
	return;
    }

    if ((err = snd_pcm_channel_flush(alsa_handler, SND_PCM_CHANNEL_PLAYBACK)) < 0)
    {
	mp_msg(MSGT_AO, MSGL_ERR, MSGTR_AO_ALSA5_FlushError, snd_strerror(err));
	return;
    }

    if ((err = snd_pcm_close(alsa_handler)) < 0)
    {
	mp_msg(MSGT_AO, MSGL_ERR, MSGTR_AO_ALSA5_PcmCloseError, snd_strerror(err));
	return;
    }
}

/* stop playing and empty buffers (for seeking/pause) */
static void reset(void)
{
    int err;

    if ((err = snd_pcm_playback_drain(alsa_handler)) < 0)
    {
	mp_msg(MSGT_AO, MSGL_ERR, MSGTR_AO_ALSA5_ResetDrainError, snd_strerror(err));
	return;
    }

    if ((err = snd_pcm_channel_flush(alsa_handler, SND_PCM_CHANNEL_PLAYBACK)) < 0)
    {
	mp_msg(MSGT_AO, MSGL_ERR, MSGTR_AO_ALSA5_ResetFlushError, snd_strerror(err));
	return;
    }

    if ((err = snd_pcm_channel_prepare(alsa_handler, SND_PCM_CHANNEL_PLAYBACK)) < 0)
    {
	mp_msg(MSGT_AO, MSGL_ERR, MSGTR_AO_ALSA5_ResetChanPrepareError, snd_strerror(err));
	return;
    }
}

/* stop playing, keep buffers (for pause) */
static void audio_pause(void)
{
    int err;

    if ((err = snd_pcm_playback_drain(alsa_handler)) < 0)
    {
	mp_msg(MSGT_AO, MSGL_ERR, MSGTR_AO_ALSA5_PauseDrainError, snd_strerror(err));
	return;
    }

    if ((err = snd_pcm_channel_flush(alsa_handler, SND_PCM_CHANNEL_PLAYBACK)) < 0)
    {
	mp_msg(MSGT_AO, MSGL_ERR, MSGTR_AO_ALSA5_PauseFlushError, snd_strerror(err));
	return;
    }
}

/* resume playing, after audio_pause() */
static void audio_resume(void)
{
    int err;
    if ((err = snd_pcm_channel_prepare(alsa_handler, SND_PCM_CHANNEL_PLAYBACK)) < 0)
    {
	mp_msg(MSGT_AO, MSGL_ERR, MSGTR_AO_ALSA5_ResumePrepareError, snd_strerror(err));
	return;
    }
}

/*
    plays 'len' bytes of 'data'
    returns: number of bytes played
*/
static int play(void* data, int len, int flags)
{
    int got_len;
    
    if (!len)
	return(0);
    
    if ((got_len = snd_pcm_write(alsa_handler, data, len)) < 0)
    {
	if (got_len == -EPIPE) /* underrun? */
	{
	    mp_msg(MSGT_AO, MSGL_ERR, MSGTR_AO_ALSA5_Underrun);
	    if ((got_len = snd_pcm_channel_prepare(alsa_handler, SND_PCM_CHANNEL_PLAYBACK)) < 0)
	    {
		mp_msg(MSGT_AO, MSGL_ERR, MSGTR_AO_ALSA5_PlaybackPrepareError, snd_strerror(got_len));
		return(0);
	    }
	    if ((got_len = snd_pcm_write(alsa_handler, data, len)) < 0)
	    {
		mp_msg(MSGT_AO, MSGL_ERR, MSGTR_AO_ALSA5_WriteErrorAfterReset,
		    snd_strerror(got_len));
		return(0);
	    }
	    return(got_len); /* 2nd write was ok */
	}
	mp_msg(MSGT_AO, MSGL_ERR, MSGTR_AO_ALSA5_OutPutError, snd_strerror(got_len));
	return(0);
    }
    return(got_len);
}

/* how many byes are free in the buffer */
static int get_space(void)
{
    snd_pcm_channel_status_t ch_stat;
    
    ch_stat.channel = SND_PCM_CHANNEL_PLAYBACK;

    if (snd_pcm_channel_status(alsa_handler, &ch_stat) < 0)
	return(0); /* error occurred */
    else
	return(ch_stat.free);
}

/* delay in seconds between first and last sample in buffer */
static float get_delay(void)
{
    snd_pcm_channel_status_t ch_stat;
    
    ch_stat.channel = SND_PCM_CHANNEL_PLAYBACK;
    
    if (snd_pcm_channel_status(alsa_handler, &ch_stat) < 0)
	return((float)ao_data.buffersize/(float)ao_data.bps); /* error occurred */
    else
	return((float)ch_stat.count/(float)ao_data.bps);
}